As more complex voice and data solutions are implemented, there is a growing desire to receive increased functionality. A trend in recent years is to provide telephony services over the internet. This is commonly referred to as Voice-over Internet Protocol or (VoIP). A protocol used to implement VoIP is Session Initiated Protocol (SIP). SIP is an Internet Engineering Task Force (IETF®)-defined signaling protocol, widely used for controlling multimedia communication sessions such as voice and video calls over IP (IETF is a trademark of The Trustees of the IETF TRUST in the United States and/or other countries). The protocol can be used for creating, modifying, and terminating two-party (i.e., unicast) or multiparty (i.e., multicast) sessions consisting of one or several media streams. The modification can involve changing addresses or ports, inviting more participants, and adding or deleting media streams. Under SIP nomenclature, placing of a call is a type of SIP transaction commonly referred to as a SIP INVITE. Typically, the SIP INVITE includes a “TO:” header that lists the destination telephone number. Since VoIP is still an emerging technology, there exists a need for more robust functionality, and leveraging of SIP technology.